[Asterisk-Users] SIP to IAX - forcing codec pass thru
Peter Gradwell
peter at gradwell.com
Mon May 22 04:49:32 MST 2006
Mark Phillips wrote:
> Hi Peter,
>
> I don't see any codec allow=blah statements. If your end user has
> something like
>
> [gradwell]
> disallow=all
> allow=gsm
>
> Then you'll be forced to send them a GSM coded call.
>
> Why not force the codec at your end by only supporting one? If the
> customer then transcodes the call when it gets forwarded to his handset
> there's not much you can do about that but at least you'll have handed
> the call off in the best format you can source.
mmm, but as you've seen, some customers like using multiple codecs. The
cisco kit is able to support a raft of options - and it does transcoding
very nicely - so the optimum solution is to have the cisco + customer's
asterisk agree on the same codec, and then have our asterisk server (in
the middle) do as little as possible.
cheers
peter
--
peter gradwell. gradwell dot com Ltd. http://www.gradwell.com/
-- engineering & hosting services for email, web and voip --
-- http://www.peter.me.uk/ -- http://www.voip.org.uk/ --
More information about the asterisk-users
mailing list